tg2sip/webrtc_dsp/modules/audio_processing/noise_suppression_impl.cc

212 lines
6.0 KiB
C++

/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "modules/audio_processing/noise_suppression_impl.h"
#include "modules/audio_processing/audio_buffer.h"
#include "rtc_base/checks.h"
#include "rtc_base/constructormagic.h"
#if defined(WEBRTC_NS_FLOAT)
#include "modules/audio_processing/ns/noise_suppression.h"
#define NS_CREATE WebRtcNs_Create
#define NS_FREE WebRtcNs_Free
#define NS_INIT WebRtcNs_Init
#define NS_SET_POLICY WebRtcNs_set_policy
typedef NsHandle NsState;
#elif defined(WEBRTC_NS_FIXED)
#include "modules/audio_processing/ns/noise_suppression_x.h"
#define NS_CREATE WebRtcNsx_Create
#define NS_FREE WebRtcNsx_Free
#define NS_INIT WebRtcNsx_Init
#define NS_SET_POLICY WebRtcNsx_set_policy
typedef NsxHandle NsState;
#endif
namespace webrtc {
class NoiseSuppressionImpl::Suppressor {
public:
explicit Suppressor(int sample_rate_hz) {
state_ = NS_CREATE();
RTC_CHECK(state_);
int error = NS_INIT(state_, sample_rate_hz);
RTC_DCHECK_EQ(0, error);
}
~Suppressor() { NS_FREE(state_); }
NsState* state() { return state_; }
private:
NsState* state_ = nullptr;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(Suppressor);
};
NoiseSuppressionImpl::NoiseSuppressionImpl(rtc::CriticalSection* crit)
: crit_(crit) {
RTC_DCHECK(crit);
}
NoiseSuppressionImpl::~NoiseSuppressionImpl() {}
void NoiseSuppressionImpl::Initialize(size_t channels, int sample_rate_hz) {
rtc::CritScope cs(crit_);
channels_ = channels;
sample_rate_hz_ = sample_rate_hz;
std::vector<std::unique_ptr<Suppressor>> new_suppressors;
if (enabled_) {
new_suppressors.resize(channels);
for (size_t i = 0; i < channels; i++) {
new_suppressors[i].reset(new Suppressor(sample_rate_hz));
}
}
suppressors_.swap(new_suppressors);
set_level(level_);
}
void NoiseSuppressionImpl::AnalyzeCaptureAudio(AudioBuffer* audio) {
RTC_DCHECK(audio);
#if defined(WEBRTC_NS_FLOAT)
rtc::CritScope cs(crit_);
if (!enabled_) {
return;
}
RTC_DCHECK_GE(160, audio->num_frames_per_band());
RTC_DCHECK_EQ(suppressors_.size(), audio->num_channels());
for (size_t i = 0; i < suppressors_.size(); i++) {
WebRtcNs_Analyze(suppressors_[i]->state(),
audio->split_bands_const_f(i)[kBand0To8kHz]);
}
#endif
}
void NoiseSuppressionImpl::ProcessCaptureAudio(AudioBuffer* audio) {
RTC_DCHECK(audio);
rtc::CritScope cs(crit_);
if (!enabled_) {
return;
}
RTC_DCHECK_GE(160, audio->num_frames_per_band());
RTC_DCHECK_EQ(suppressors_.size(), audio->num_channels());
for (size_t i = 0; i < suppressors_.size(); i++) {
#if defined(WEBRTC_NS_FLOAT)
WebRtcNs_Process(suppressors_[i]->state(), audio->split_bands_const_f(i),
audio->num_bands(), audio->split_bands_f(i));
#elif defined(WEBRTC_NS_FIXED)
WebRtcNsx_Process(suppressors_[i]->state(), audio->split_bands_const(i),
audio->num_bands(), audio->split_bands(i));
#endif
}
}
int NoiseSuppressionImpl::Enable(bool enable) {
rtc::CritScope cs(crit_);
if (enabled_ != enable) {
enabled_ = enable;
Initialize(channels_, sample_rate_hz_);
}
return AudioProcessing::kNoError;
}
bool NoiseSuppressionImpl::is_enabled() const {
rtc::CritScope cs(crit_);
return enabled_;
}
int NoiseSuppressionImpl::set_level(Level level) {
int policy = 1;
switch (level) {
case NoiseSuppression::kLow:
policy = 0;
break;
case NoiseSuppression::kModerate:
policy = 1;
break;
case NoiseSuppression::kHigh:
policy = 2;
break;
case NoiseSuppression::kVeryHigh:
policy = 3;
break;
default:
RTC_NOTREACHED();
}
rtc::CritScope cs(crit_);
level_ = level;
for (auto& suppressor : suppressors_) {
int error = NS_SET_POLICY(suppressor->state(), policy);
RTC_DCHECK_EQ(0, error);
}
return AudioProcessing::kNoError;
}
NoiseSuppression::Level NoiseSuppressionImpl::level() const {
rtc::CritScope cs(crit_);
return level_;
}
float NoiseSuppressionImpl::speech_probability() const {
rtc::CritScope cs(crit_);
#if defined(WEBRTC_NS_FLOAT)
float probability_average = 0.0f;
for (auto& suppressor : suppressors_) {
probability_average +=
WebRtcNs_prior_speech_probability(suppressor->state());
}
if (!suppressors_.empty()) {
probability_average /= suppressors_.size();
}
return probability_average;
#elif defined(WEBRTC_NS_FIXED)
// TODO(peah): Returning error code as a float! Remove this.
// Currently not available for the fixed point implementation.
return AudioProcessing::kUnsupportedFunctionError;
#endif
}
std::vector<float> NoiseSuppressionImpl::NoiseEstimate() {
rtc::CritScope cs(crit_);
std::vector<float> noise_estimate;
#if defined(WEBRTC_NS_FLOAT)
const float kNumChannelsFraction = 1.f / suppressors_.size();
noise_estimate.assign(WebRtcNs_num_freq(), 0.f);
for (auto& suppressor : suppressors_) {
const float* noise = WebRtcNs_noise_estimate(suppressor->state());
for (size_t i = 0; i < noise_estimate.size(); ++i) {
noise_estimate[i] += kNumChannelsFraction * noise[i];
}
}
#elif defined(WEBRTC_NS_FIXED)
noise_estimate.assign(WebRtcNsx_num_freq(), 0.f);
for (auto& suppressor : suppressors_) {
int q_noise;
const uint32_t* noise =
WebRtcNsx_noise_estimate(suppressor->state(), &q_noise);
const float kNormalizationFactor =
1.f / ((1 << q_noise) * suppressors_.size());
for (size_t i = 0; i < noise_estimate.size(); ++i) {
noise_estimate[i] += kNormalizationFactor * noise[i];
}
}
#endif
return noise_estimate;
}
size_t NoiseSuppressionImpl::num_noise_bins() {
#if defined(WEBRTC_NS_FLOAT)
return WebRtcNs_num_freq();
#elif defined(WEBRTC_NS_FIXED)
return WebRtcNsx_num_freq();
#endif
}
} // namespace webrtc